Subject: This knowledge article discusses information provided by the Orum Engineering Team regarding recommended networking configurations to best sync with the product including protocols and ports. Please create a ticket for our Customer Support Team if further assistance or explanation is required. Thank you!
In terms of Orum's networking, users first connect via WebSocket to our SIP server. Because we're establishing an audio connection with your local machine, we have to perform NAT traversal. To do so, your computer will connect to Google's public STUN servers to get its external IP address and port. This IP and port is communicated to our SIP server, and then audio data is sent via RTP (which operates over UDP) between our SIP server and your local machine. If you have a symmetric NAT, or a firewall dropping UDP packets, it can cause issues here.
Here are the protocols and ports:
Orum's SIP server: siphon.orum.com at IP: 22.214.171.124
For RTP, you need to allow outbound UDP to the above server for ports in range 10000 - 65535, and any response packets shouldn't be dropped (i.e. this is symmetric RTP).
The websocket connection upgrades from standard HTTPS to the above server.
In order for your users to determine their external IP address, you need to allow outbound UDP connections to Google's public STUN servers:
We'd recommend whitelisting domains rather than IPs if possible, since IPs can change or be added over time (both for us and for Google). Your users also need to allow microphone access within their browser to use the in-browser dialer.